Packet mode communication networks transmit information/data in the form of packets. One example of such a network is the Internet network which is operated with IP protocol (Internet protocol). IP is used for transmitting data from higher-level protocols, such as TCP and UDP, in IP packets from one host to another host in the network. Each packet is handled independent of other packets and each packet may reach the destination through different network routes. The communication services, such as voice/sound or visual communications, require a certain level of quality across the IP network.
One prior art approach of determining quality for packetized information is shown in WO 2005/004370, which discloses a method for near real time analysis which samples packets from a stream of IP packets that represent a communication session between a pair of endpoints and determines two metrics from the sampled packets, quantity of lost packets and packet timing. The packet loss is calculated by looking at the RTP sequence number in each packet. Gaps in the sequence represent lost packets and the packet loss is calculated as the number of lost packets divided by the sum of received packets plus lost packets.
However, this prior art approach measures the quality of the network in a complicated manner involving several metrics to consider. Additionally, protocols like RTP require large headers of the packets, typically 12 octets for the RTP protocol, which is capacity consuming. Other existing solutions in this area involve hardware (HW) that is costly.
There is, thus, today no cheap, easy way for two peer endpoints of an IP transmission network to know the overall transmission quality of the intermediary IP network connecting them.